In applications, such as video/data conferencing applications employing Voice over IP (VoIP), the timely delivery of real-time media packets may be valued above reliability levels provided via retransmission of non-received packets. Retransmission of packets not received by a receiving network device is provided in implementations of TCP. One approach to this issue is to use User Datagram Protocol (UDP) instead of TCP for the delivery of real-time media. However, many networks employ firewalls (or other middleboxes) that block the usage of UDP ports. Thus, a real-time video/data conferencing application may be forced to use only TCP.
The forced usage of TCP for real-time video/data conferencing applications results in the retransmission of each real-time media packet dropped between a transmitting network device and a receiving network device. Each retransmission results in a minimum delay of one round trip time (RTT). Such delays are unacceptable to provide the performance desired by real-time video/data conferencing applications as the RTTs may be quite long relative to the nominal media packet inter-departure times.